FIG. 1A, 1B, 1C shows three example network communication configurations. In FIG. 1A, the calling party initiates a call over the public switch telephone network (PSTN) 30 using either a wireline telephone 20 or wireless telephone (e.g., mobile telephone) 10. The call is routed through a gateway 50 to the Internet 60 to save on long distance toll costs. At the remote end, the call is routed to the gateway 50 and PSTN 30 nearest to the called party who may be using either a wireline 20 or wireless phone 10. If the called party does not answer the call, the call is diverted to voice mail 40. Voice mail 40 answers your call, records a message, and notifies you when you have new messages. Note, a wireless telephone call 10 is set-up, managed and routed to the PSTN 30 by the mobile switch center (not shown).
In FIG. 1B, the calling party initiates a call over the PSTN 30 using either a wireline telephone 20 or wireless telephone 10. In this example, the call is routed solely through the PSTN 30 to the remote end where the called party may be using either a wireline 20 or wireless telephone 10. If the called party does not answer the call but pays for voice mail 40 service as part of their communications bill, the call is diverted to voice mail 40.
In FIG. 1C, the calling party may initiate a call over the PSTN 30 using either a wireline telephone 20 or wireless telephone 10. Similarly, the calling party may initiate a call over the Internet 60 using a H.323 terminal or the session initiation protocol (SIP) telephone 70. H.323/SIP terminals include IP telephones 70 or IP softphones 80. An IP telephone provides communications capability as analog or digital telephones provide except that communications are routed via the Internet or data network rather than via PSTN. An IP softphone 80 is a client-based telephony application for the desktop PC 80 or laptop 80 that has similar functionality as a desktop IP telephone 70.
At the remote end, the private branch exchange (PBX) 36 is a telephone system that supports enterprise users (college, government office, business, etc.) by answering and transferring inbound and outbound telephone calls to and from the PSTN 30 or Internet 60. All enterprise users share external telephone lines, i.e., trunk lines 35, which saves the cost of requiring a line for each user to the telephone company's central office (CO) (not shown but part of the PSTN 30). PBXs 36 have evolved from being proprietary hardware/software systems completely separate from the packet switched network or LAN 39 to systems running on off-the-shelf servers, interoperable with other servers through open standards and communicating via the LAN 39. Furthermore, the PBX 36 has evolved from strictly routing local and long distance telephone calls over the PSTN 30 to additionally providing the capability to route local and long distance telephone calls over the Internet 60 or over the LAN 39. The PBX 36 operating on packet switched networks allow the enterprise to reduce costs by maintaining one network instead of two (the data and telephone) and reducing charges from toll calls by routing some calls over the packet switched network 39 or Internet 60.
As shown generally in FIGS. 1A, 1B, 1C and 2 when a call is initiated using a wireless telephone 10 or a wireline telephone 20, 41, 42 from outside or within an enterprise and sent via the PSTN 30, the calling party 250 dials the telephone number which is transmitted to the nearest central office (CO) (part of PSTN 52, FIG. 2) along with a call set-up message (FIG. 2, 254). The local exchange CO comprises one or more carrier class switches (not shown), which take calls and routes the calls to the proper destination based on the dialed number. If the call is a long distance call, the call goes from the local CO to a long distance carrier's class switch on route to the local CO nearest the called party. The call set-up message (FIG. 2, 254; FIG. 3, 371) for a call sent over the various switches is governed by standards, which are particular to the network in use.
As shown in FIG. 2, the local CO (which is part of the PSTN 52) locates the remote end-point 253 using the dialed telephone number. Once the CO locates the remote end-point 253, the CO sends a call set-up message 254a to the remote end-point 253 as shown in FIG. 2. If the remote endpoint 253 is on and not busy (FIG. 3, 372), an alerting (ringing) signal 256a is sent back to the CO and the CO forwards the alerting (ringing) signal 256 to the calling party 250. The alerting signal tells the calling party the remote endpoint 253 has not answered the call. As shown in FIG. 3, once the called party goes off-hook, i.e., answers the telephone, the telephone at the remote endpoint (FIG. 2, 253) sends a call-connect message (FIG. 3, 357) to the CO. The CO forwards the call-connect message 257 to the calling party (FIG. 2, 251). Now the two endpoints will begin transmitting voice or data between them over the PSTN (FIG. 3, 376, 377). A call termination message is sent by either the calling party or called party to disconnect, i.e., terminate the call (FIG. 2, 258, 258a, 259a, 259; FIG. 3, 378).
As shown in FIG. 2, the remote endpoint maybe unavailable to answer the call. That is, the telephone is turned off, not connected, or busy, or the called party simply does not answer the call (FIG. 2, 262a; FIG. 3, 372). In this case, if the called party does not have a “follow-me” service (FIG. 3, 373), the call is diverted to a multi-media message storage system (e.g., voice mail system) and the call is answered by the multi-media message storage system (FIG. 2, 263a, 264a, 264). Once the calling party records the voicemail message (FIG. 3, 374), the calling party terminates the call (FIG. 2, 265, 265a, 266, 266a).
If the called party has a “follow-me” service (FIG. 3, 373), the CO diverts the call to a pre-defined next location (FIG. 3, 373) and tries to contact the called party at each “follow-me” number administered for called party. If the called party is not available at any of the “follow-me” numbers, the CO diverts the call to a multi-media storage system 274.
For IP desktop telephones (FIG. 1, 70) or softphones (FIG. 1, 80) sending calls via the Internet, using session initiation protocol (SIP) or H.323 protocol, the messages sent between the end-points to establish a call and set-up a communication channel are governed by the particular protocol in use. Regardless, the called party at the remote end-point is alerted to an incoming call by a standard ringing signal if the IP end-point is available (FIG. 3, 372). Once the call is answered by the called party, the appropriate establishment and connection messages (FIG. 3, 375) are sent via the Internet or over the data network. Otherwise the call is similarly diverted to a pre-defined next location (FIG. 3, 373), such as a “follow-me” number as administered in the service preferences, or a voice mail system (FIG. 3, 374).
Unfortunately, neither wireline or wireless telephone systems or even paging services currently allow the calling party to send a recorded or real-time data/bearer stream including voice, text, images or video attached to a call set-up message to initiate a telephone call and to provide information to the called party before the called party answers the telephone. For example, mobile phones enabled to send and receive SMS or MMS, the short message service (SMS) protocol allows mobile users to send short text messages and the multimedia message service (MMS) protocol allows mobile users to send multimedia messages. SMS also allows a mobile user to send short text messages to and receive text messages from email, paging services or informational services (such as receiving stock quotes). MMS adds images, text, audio clips and video to SMS messages. However, both SMS and MMS messages are not delivered in real time and hence cannot initiate a telephone call.
The mobile switch center (not shown) sends SMS messages to a mobile message service center (not shown). If the mobile phone user is available, the SMS message is immediately deliverable to the recipient and the mobile message center sends the message to the recipient. Otherwise, the message is stored in the mobile message service center until the mobile user is available.
The mobile switch center (not shown) sends MMS messages to a mobile message service center (not shown). The message service center sends the sender a message confirmation that the message was sent. The message service center then sends the recipient a message notification that a new message has arrived. The recipient can download the message immediately or later. Once the recipient has successfully downloaded the message, the sender gets a message delivered confirmation message.
For mobile phones that use polyphonic ring tone technology, the mobile telephone user can download various high quality tones and administer their user preferences to play a particular tone when a particular incoming call arrives. Alternatively, the mobile telephone user can record their voice and administer their user preferences to play the recording of their voice when a particular person is calling or for any incoming call. Note, these polyphonic ring tones are administered and recorded by the called party mobile telephone owner to play when a call is received from a calling party.
Current voice paging systems associated with telephones are typically used in a facility to broadcast messages to locate individuals or announce messages, such as emergencies or sales. When the paging system is used to locate individuals, the individual still needs to call back the person who initiated the page. The person initiating the page cannot use the paging system to initiate a telephone call to locate an individual. The person initiating the page cannot send a real-time data/bearer stream attached to a call set-up message to initiate a call and alert a called party before the called party answers the call.